Is it possible to get a wireshark of the call at either end. Acrobits
probably cannot be configured for the media ports, but not being able to see
why rtp doesnt establish just leads to conjecture.

Have you tried to get a siptrace of the call?

I am thinking the wireshark or siptrace will give a clue as to how "late" in
the transaction the ack is being received for the media to establish. If it
works on wifi, but not on 3g, I think it's safe to assume the carrier is
delaying it or changing the source ports (transparent proxy). Timing is all
important to media.

On Tue, May 11, 2010 at 10:16 AM, Stephen D. Miller
<steph...@thinkgear.com>wrote:

>
> Given that the Acrobits softphone registers with the server successfully
> and receives responses from the INVITE, I think it's safe to assume 5060
> isn't being blocked by AT&T in this area.
>
> All call-handling related traffic (REGISTER, INVITE, etc) appears to be
> flowing between the endpoints correctly---just the UDP RTP stream isn't
> getting established.  And the direction that it's failing (from local
> phone to remote) isn't the direction I'd expect it to...
>
> What I don't understand is how does the local SIP phone knows (or is
> supposed to know) that the destination of the outbound RTP stream is
> remote (non-LAN and behind a NAT) so that it can address packets
> accordingly?
>
> sdm
>
> > -----Original Message-----
> > From: Nathaniel Watkins [mailto:nwatk...@garrettcounty.org]
> > Sent: Tuesday, May 11, 2010 10:13 AM
> > To: Stephen D. Miller; sipx-users@list.sipfoundry.org
> > Subject: RE: Another remote worker configuration question...
> >
> > Is you provider blocking port 5060?  Might consider
> > connecting via vpn first?
> >
> > -----Original Message-----
> > From: sipx-users-boun...@list.sipfoundry.org
> > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
> > Stephen D. Miller
> > Sent: Tuesday, May 11, 2010 10:02 AM
> > To: sipx-users@list.sipfoundry.org
> > Subject: [sipx-users] Another remote worker configuration question...
> >
> >
> > I have a newly installed sipxecs 4.2 system (replaced a
> > previously installed v3.10 system).  System is behind a NAT
> > firewall.  All works as expected (internal extension to
> > extension calls, outbound calls, inbound calls, etc). I have
> > also followed the v4.2 remote worker config guide and
> > configured the server accordingly (port forwarding, local
> > subnet specified, etc).
> >
> > I have installed Acrobits SIP client on an iPhone and
> > configured it as an extension in sipxecs.  Using the local
> > WiFi network, the iPhone SIP client registers and can
> > call/communicate with any extension as expected.
> >
> > Using 3G, the Acrobits SIP client registers properly and can
> > call internal extensions, but the media streams (RTP) are
> > never established.
> >
> > I have captured the net traffic at the server in the failure
> > case and it appears that the RTP traffic inbound from the
> > (remote) iPhone is arriving at the sipxecs server and is
> > being properly sent on to the internal extension.  However, I
> > don't see any RTP traffic heading outbound to the iPhone.  I
> > think this is the problem...
> >
> > Does anyone have recommendations for further diagnosing this scenario?
> > I can post a wireshark capture of the traffic in the failure
> > case...if that's helpful.
> >
> > Thanks in advance-
> >
> > sdm
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